Inter-personal and inter-organizational communication has increased in complexity. For example, the public switched telephone network (PSTN) currently includes separate networks to accommodate different traffic types. Command and control functionality of the PSTN is built around a connection-oriented class structure. Call processing is completed by a complex array of switches, processors, and control networks, such as the Signaling System 7 (SS7). The PSTN is built on a class structure of specialized devices that perform single-purpose functions. For example, Class 5 switches provide local access and call waiting; Class 4 switches provide long distance, toll calls and billing. Connectivity is also connection-oriented, where two devices must establish a discrete, deterministic connection or voice channel before communication can begin. The voice channel also must remain in place for the duration of the call. The PSTN is also a hierarchical one-to-many network that may result in traffic congestion, or blocking, thus creating the need for centralized control to proactively re-route the traffic. New service and feature development on this complex network is time-consuming and difficult, requiring a great deal of development time and resources.
In contrast to the PSTN, an Internet Protocol (IP) network, such as the Internet, uses a packet-based architecture, which is generally more economical than circuit switched networks (such as SS7). In an IP network, every packet of information carries all the necessary control information needed to connect the originating device to its destination using only the resources needed. This allows for uncomplicated (less) throttle points, geographic expansion, and transport integration. An Internet Protocol (IP) network is typically used to handle the large amounts of time-sensitive data such as telephony data and/or video or video-on-demand data.
In addition, a variety of wireless networks has been developed, each with its own protocol. As these developments have occurred, users increasingly desire voice as well as IP data to be communicated over wireless networks, such as time division multiplex access (TDMA), call division multiplex access (CDMA), advanced mobile phone service (AMPS), and wideband and proposed third generation (3G) wireless systems W-CDMA, W-TDMA, and others, which provide for IP data as well as voice transfer.
An H.323 protocol environment or Session Initiation Protocol (SIP) provides a way for transmitting voice communications within IP networks. In general, H.323 define a set of call control, channel set-up, and code C specifications for transmitting real-time voice and video over networks that do not offer guaranteed service or quality of service—such as packet networks, and in particular, the Internet, local area networks, wide area networks, and Intranets. SIP is a session-layered control (signaling) protocol for creating, modifying and terminating session with one or more participants.
Communications of telephone calls between the PSTN (which uses SS7 protocol), wireless and IP networks (which additionally use H.323 protocol to transfer voice over the IP) have traditionally used complicated signaling gateways to perform conversions between the PSTN signaling functions and the Net Protocol Network signaling functions. One function that is performed in facilitating communication between various types of networks is determining through which route a telephone call will take place in the network in which the call will terminate. For the PSTN, the route a telephone call takes place is comprised of a plurality of trunks. A trunk (not specifically shown) is a communication line between two switching systems, such as between a central office and a private branch exchange (PBX).
Routing time-sensitive data such as telephone calls whether between networks of the same type or between different networks, typically requires complex signaling algorithms to be utilized.
Furthermore, the number of resources required to maintain such a network and to route these calls is burdensome. For example, in order to complete a call, the system must “dip down” into both a calling party's and destination party's Automatic Number Identification (ANI) database. This process is performed to insure that subscriber information such as caller ID, call blocking, and others, are correctly obtained for each of the parties to properly insure completion of the call. Such processing is inefficient because, in part, many circuits are tied up during this process. This “dip down” process reaches below the physical layer to perform authenticating and routing functions. For example, a single call session may require numerous (on the order of hundreds) traversals to and accesses to one or more databases. Such processing also limits the scalability of systems. These numerous database accesses are required, for examine, when mechanically based circuits translate each digit as it is dialed in. In some cases, this process may involve local, regional, and/or national tandem switching elements in the transit network, each of which has an ANI table. Each of these ANI databases are typically maintained by separate management information systems (MIS). Processing wireless calls only increases the complexity of such systems. For example, routing, authentication and verification, and fraud detection require information such as, but not limited to, home location register (HLR) and/or visitor location register (VLR), in addition to ANI information to complete a call. Additional information is also required with mobile devices such as cellular phones or radios, which typically utilize a device identifier such as a mobile identification number (MIN), an electronic serial number (ESN), or an international mobile subscriber identity (IMSI). These identifiers are required to identify both a device and subscriber to a network for purposes of authentication, tracking, and billing. Moreover, fraud detection typically requires information such as correlation tables to be used in conjunction with the IMSI to verify subscriber information and/or services that are provided to an origination or termination party. Thus, this process of obtaining information for each call may be extremely complex and consume a great deal of time and resources while the information is being obtained. Moreover, congestion, delay, or malfunction within an IP network is typically handled by simply re-sending data packets to a terminating party. Such retransmissions decrease network efficiency by tying up circuits that may be inoperative and provide no network self-awareness.